webrtc/modules/audio_processing/aec3/mock/mock_render_delay_buffer.cc
Gustaf Ullberg d3ead1a942 AEC3: 'Block' class
This change adds a Block class to reduce the need for std::vector<std::vector<std::vector<float>>>. This make the code
easier to read and less error prone.

It also enables future changes to the underlying data structure of a
block. For instance, the data of all bands and channels could be stored
in a single vector.

The change has been verified to be bit-exact.

Bug: webrtc:14089
Change-Id: Ied9a78124c0bbafe0e912017aef91f7c311de2ae
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262252
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36968}
2022-05-23 09:53:46 +00:00

36 lines
1.4 KiB
C++

/*
* Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include "modules/audio_processing/aec3/mock/mock_render_delay_buffer.h"
namespace webrtc {
namespace test {
MockRenderDelayBuffer::MockRenderDelayBuffer(int sample_rate_hz,
size_t num_channels)
: block_buffer_(GetRenderDelayBufferSize(4, 4, 12),
NumBandsForRate(sample_rate_hz),
num_channels),
spectrum_buffer_(block_buffer_.buffer.size(), num_channels),
fft_buffer_(block_buffer_.buffer.size(), num_channels),
render_buffer_(&block_buffer_, &spectrum_buffer_, &fft_buffer_),
downsampled_render_buffer_(GetDownSampledBufferSize(4, 4)) {
ON_CALL(*this, GetRenderBuffer())
.WillByDefault(
::testing::Invoke(this, &MockRenderDelayBuffer::FakeGetRenderBuffer));
ON_CALL(*this, GetDownsampledRenderBuffer())
.WillByDefault(::testing::Invoke(
this, &MockRenderDelayBuffer::FakeGetDownsampledRenderBuffer));
}
MockRenderDelayBuffer::~MockRenderDelayBuffer() = default;
} // namespace test
} // namespace webrtc